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metrics.py
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'''
Reference:
https://github.com/santi-pdp/segan_pytorch/blob/master/segan/utils.py
https://github.com/svj1991/Adaptive_front_ends/blob/master/sepcosts.py
'''
from subprocess import run, PIPE
from scipy.linalg import toeplitz
from scipy.io import wavfile
# from numba import jit, int32, float32
import soundfile as sf
from scipy.signal import lfilter
from scipy.interpolate import interp1d
import glob
import numpy as np
import tempfile
import os
import re
import torch
import torch.nn as nn
import torch.nn.functional as F
from stft import *
EPS = 1e-8
def eval_composite(clean_utt, Genh_utt, noisy_utt=None):
clean_utt = clean_utt.reshape(-1)
Genh_utt = Genh_utt.reshape(-1)
csig, cbak, covl, pesq, ssnr = CompositeEval(clean_utt,
Genh_utt,
True)
evals = {'csig':csig, 'cbak':cbak, 'covl':covl,
'pesq':pesq, 'ssnr':ssnr}
if noisy_utt is not None:
noisy_utt = noisy_utt.reshape(-1)
csig, cbak, covl, \
pesq, ssnr = CompositeEval(clean_utt,
noisy_utt,
True)
return evals, {'csig':csig, 'cbak':cbak, 'covl':covl,
'pesq':pesq, 'ssnr':ssnr}
else:
return evals
def PESQ(ref_wav, deg_wav):
# reference wav
# degraded wav
tfl = tempfile.NamedTemporaryFile()
ref_tfl = tfl.name + '_ref.wav'
deg_tfl = tfl.name + '_deg.wav'
sf.write(ref_tfl, ref_wav, 16000, subtype='PCM_16')
sf.write(deg_tfl, deg_wav, 16000, subtype='PCM_16')
curr_dir = os.getcwd()
# Write both to tmp files and then eval with pesqmain
try:
p = run(['./pesq_src/PESQ',
ref_tfl, deg_tfl, '+16000', '+wb'],
stdout=PIPE,
encoding='ascii')
res_line = p.stdout.split('\n')[-2]
results = re.split('\s+', res_line)
return results[-1]
except FileNotFoundError:
print('PESQ not found! Please add it your PATH')
def SSNR(ref_wav, deg_wav, srate=16000, eps=1e-10):
""" Segmental Signal-to-Noise Ratio Objective Speech Quality Measure
This function implements the segmental signal-to-noise ratio
as defined in [1, p. 45] (see Equation 2.12).
"""
clean_speech = ref_wav
processed_speech = deg_wav
clean_length = ref_wav.shape[0]
processed_length = deg_wav.shape[0]
# scale both to have same dynamic range. Remove DC too.
dif = ref_wav - deg_wav
overall_snr = 10 * np.log10(np.sum(ref_wav ** 2) / (np.sum(dif ** 2) +
10e-20))
# global variables
winlength = int(np.round(30 * srate / 1000)) # 30 msecs
skiprate = winlength // 4
MIN_SNR = -10
MAX_SNR = 35
# For each frame, calculate SSNR
num_frames = int(clean_length / skiprate - (winlength/skiprate))
start = 0
time = np.linspace(1, winlength, winlength) / (winlength + 1)
window = 0.5 * (1 - np.cos(2 * np.pi * time))
segmental_snr = []
for frame_count in range(int(num_frames)):
# (1) get the frames for the test and ref speech.
# Apply Hanning Window
clean_frame = clean_speech[start:start+winlength]
processed_frame = processed_speech[start:start+winlength]
clean_frame = clean_frame * window
processed_frame = processed_frame * window
# (2) Compute Segmental SNR
signal_energy = np.sum(clean_frame ** 2)
noise_energy = np.sum((clean_frame - processed_frame) ** 2)
segmental_snr.append(10 * np.log10(signal_energy / (noise_energy + eps)+ eps))
segmental_snr[-1] = max(segmental_snr[-1], MIN_SNR)
segmental_snr[-1] = min(segmental_snr[-1], MAX_SNR)
start += int(skiprate)
return overall_snr, segmental_snr
def CompositeEval(ref_wav, deg_wav, log_all=False):
# returns [sig, bak, ovl]
alpha = 0.95
len_ = min(ref_wav.shape[0], deg_wav.shape[0])
ref_wav = ref_wav[:len_]
ref_len = ref_wav.shape[0]
deg_wav = deg_wav[:len_]
# Compute WSS measure
wss_dist_vec = wss(ref_wav, deg_wav, 16000)
wss_dist_vec = sorted(wss_dist_vec, reverse=False)
wss_dist = np.mean(wss_dist_vec[:int(round(len(wss_dist_vec) * alpha))])
# Compute LLR measure
LLR_dist = llr(ref_wav, deg_wav, 16000)
LLR_dist = sorted(LLR_dist, reverse=False)
LLRs = LLR_dist
LLR_len = round(len(LLR_dist) * alpha)
llr_mean = np.mean(LLRs[:LLR_len])
# Compute the SSNR
snr_mean, segsnr_mean = SSNR(ref_wav, deg_wav, 16000)
segSNR = np.mean(segsnr_mean)
# Compute the PESQ
pesq_raw = PESQ(ref_wav, deg_wav)
if 'error!' not in pesq_raw:
pesq_raw = float(pesq_raw)
else:
pesq_raw = -1.
def trim_mos(val):
return min(max(val, 1), 5)
Csig = 3.093 - 1.029 * llr_mean + 0.603 * pesq_raw - 0.009 * wss_dist
Csig = trim_mos(Csig)
Cbak = 1.634 + 0.478 * pesq_raw - 0.007 * wss_dist + 0.063 * segSNR
Cbak = trim_mos(Cbak)
Covl = 1.594 + 0.805 * pesq_raw - 0.512 * llr_mean - 0.007 * wss_dist
Covl = trim_mos(Covl)
if log_all:
return Csig, Cbak, Covl, pesq_raw, segSNR
else:
return Csig, Cbak, Covl
def wss(ref_wav, deg_wav, srate):
clean_speech = ref_wav
processed_speech = deg_wav
clean_length = ref_wav.shape[0]
processed_length = deg_wav.shape[0]
assert clean_length == processed_length, clean_length
winlength = round(30 * srate / 1000.) # 240 wlen in samples
skiprate = np.floor(winlength / 4)
max_freq = srate / 2
num_crit = 25 # num of critical bands
USE_FFT_SPECTRUM = 1
n_fft = int(2 ** np.ceil(np.log(2*winlength)/np.log(2)))
n_fftby2 = int(n_fft / 2)
Kmax = 20
Klocmax = 1
# Critical band filter definitions (Center frequency and BW in Hz)
cent_freq = [50., 120, 190, 260, 330, 400, 470, 540, 617.372,
703.378, 798.717, 904.128, 1020.38, 1148.30,
1288.72, 1442.54, 1610.70, 1794.16, 1993.93,
2211.08, 2446.71, 2701.97, 2978.04, 3276.17,
3597.63]
bandwidth = [70., 70, 70, 70, 70, 70, 70, 77.3724, 86.0056,
95.3398, 105.411, 116.256, 127.914, 140.423,
153.823, 168.154, 183.457, 199.776, 217.153,
235.631, 255.255, 276.072, 298.126, 321.465,
346.136]
bw_min = bandwidth[0] # min critical bandwidth
# set up critical band filters. Note here that Gaussianly shaped filters
# are used. Also, the sum of the filter weights are equivalent for each
# critical band filter. Filter less than -30 dB and set to zero.
min_factor = np.exp(-30. / (2 * 2.303)) # -30 dB point of filter
crit_filter = np.zeros((num_crit, n_fftby2))
all_f0 = []
for i in range(num_crit):
f0 = (cent_freq[i] / max_freq) * (n_fftby2)
all_f0.append(np.floor(f0))
bw = (bandwidth[i] / max_freq) * (n_fftby2)
norm_factor = np.log(bw_min) - np.log(bandwidth[i])
j = list(range(n_fftby2))
crit_filter[i, :] = np.exp(-11 * (((j - np.floor(f0)) / bw) ** 2) + \
norm_factor)
crit_filter[i, :] = crit_filter[i, :] * (crit_filter[i, :] > \
min_factor)
# For each frame of input speech, compute Weighted Spectral Slope Measure
# num of frames
num_frames = int(clean_length / skiprate - (winlength / skiprate))
start = 0 # starting sample
time = np.linspace(1, winlength, winlength) / (winlength + 1)
window = 0.5 * (1 - np.cos(2 * np.pi * time))
distortion = []
for frame_count in range(num_frames):
# (1) Get the Frames for the test and reference speeech.
# Multiply by Hanning window.
clean_frame = clean_speech[start:start+winlength]
processed_frame = processed_speech[start:start+winlength]
clean_frame = clean_frame * window
processed_frame = processed_frame * window
# (2) Compuet Power Spectrum of clean and processed
clean_spec = (np.abs(np.fft.fft(clean_frame, n_fft)) ** 2)
processed_spec = (np.abs(np.fft.fft(processed_frame, n_fft)) ** 2)
clean_energy = [None] * num_crit
processed_energy = [None] * num_crit
# (3) Compute Filterbank output energies (in dB)
for i in range(num_crit):
clean_energy[i] = np.sum(clean_spec[:n_fftby2] * \
crit_filter[i, :])
processed_energy[i] = np.sum(processed_spec[:n_fftby2] * \
crit_filter[i, :])
clean_energy = np.array(clean_energy).reshape(-1, 1)
eps = np.ones((clean_energy.shape[0], 1)) * 1e-10
clean_energy = np.concatenate((clean_energy, eps), axis=1)
clean_energy = 10 * np.log10(np.max(clean_energy, axis=1))
processed_energy = np.array(processed_energy).reshape(-1, 1)
processed_energy = np.concatenate((processed_energy, eps), axis=1)
processed_energy = 10 * np.log10(np.max(processed_energy, axis=1))
# (4) Compute Spectral Shape (dB[i+1] - dB[i])
clean_slope = clean_energy[1:num_crit] - clean_energy[:num_crit-1]
processed_slope = processed_energy[1:num_crit] - \
processed_energy[:num_crit-1]
# (5) Find the nearest peak locations in the spectra to each
# critical band. If the slope is negative, we search
# to the left. If positive, we search to the right.
clean_loc_peak = []
processed_loc_peak = []
for i in range(num_crit - 1):
if clean_slope[i] > 0:
# search to the right
n = i
while n < num_crit - 1 and clean_slope[n] > 0:
n += 1
clean_loc_peak.append(clean_energy[n - 1])
else:
# search to the left
n = i
while n >= 0 and clean_slope[n] <= 0:
n -= 1
clean_loc_peak.append(clean_energy[n + 1])
# find the peaks in the processed speech signal
if processed_slope[i] > 0:
n = i
while n < num_crit - 1 and processed_slope[n] > 0:
n += 1
processed_loc_peak.append(processed_energy[n - 1])
else:
n = i
while n >= 0 and processed_slope[n] <= 0:
n -= 1
processed_loc_peak.append(processed_energy[n + 1])
# (6) Compuet the WSS Measure for this frame. This includes
# determination of the weighting functino
dBMax_clean = max(clean_energy)
dBMax_processed = max(processed_energy)
# The weights are calculated by averaging individual
# weighting factors from the clean and processed frame.
# These weights W_clean and W_processed should range
# from 0 to 1 and place more emphasis on spectral
# peaks and less emphasis on slope differences in spectral
# valleys. This procedure is described on page 1280 of
# Klatt's 1982 ICASSP paper.
clean_loc_peak = np.array(clean_loc_peak)
processed_loc_peak = np.array(processed_loc_peak)
Wmax_clean = Kmax / (Kmax + dBMax_clean - clean_energy[:num_crit-1])
Wlocmax_clean = Klocmax / (Klocmax + clean_loc_peak - \
clean_energy[:num_crit-1])
W_clean = Wmax_clean * Wlocmax_clean
Wmax_processed = Kmax / (Kmax + dBMax_processed - \
processed_energy[:num_crit-1])
Wlocmax_processed = Klocmax / (Klocmax + processed_loc_peak - \
processed_energy[:num_crit-1])
W_processed = Wmax_processed * Wlocmax_processed
W = (W_clean + W_processed) / 2
distortion.append(np.sum(W * (clean_slope[:num_crit - 1] - \
processed_slope[:num_crit - 1]) ** 2))
# this normalization is not part of Klatt's paper, but helps
# to normalize the meaasure. Here we scale the measure by the sum of the
# weights
distortion[frame_count] = distortion[frame_count] / np.sum(W)
start += int(skiprate)
return distortion
def llr(ref_wav, deg_wav, srate):
clean_speech = ref_wav
processed_speech = deg_wav
clean_length = ref_wav.shape[0]
processed_length = deg_wav.shape[0]
assert clean_length == processed_length, clean_length
winlength = round(30 * srate / 1000.) # 240 wlen in samples
skiprate = np.floor(winlength / 4)
if srate < 10000:
# LPC analysis order
P = 10
else:
P = 16
# For each frame of input speech, calculate the Log Likelihood Ratio
num_frames = int(clean_length / skiprate - (winlength / skiprate))
start = 0
time = np.linspace(1, winlength, winlength) / (winlength + 1)
window = 0.5 * (1 - np.cos(2 * np.pi * time))
distortion = []
for frame_count in range(num_frames):
# (1) Get the Frames for the test and reference speeech.
# Multiply by Hanning window.
clean_frame = clean_speech[start:start+winlength]
processed_frame = processed_speech[start:start+winlength]
clean_frame = clean_frame * window
processed_frame = processed_frame * window
# (2) Get the autocorrelation logs and LPC params used
# to compute the LLR measure
R_clean, Ref_clean, A_clean = lpcoeff(clean_frame, P)
R_processed, Ref_processed, A_processed = lpcoeff(processed_frame, P)
A_clean = A_clean[None, :]
A_processed = A_processed[None, :]
#print('A_clean shape: ', A_clean.shape)
#print('toe(R_clean) shape: ', toeplitz(R_clean).shape)
#print('A_clean: ', A_clean)
#print('A_processed: ', A_processed)
#print('toe(R_clean): ', toeplitz(R_clean))
# (3) Compute the LLR measure
numerator = A_processed.dot(toeplitz(R_clean)).dot(A_processed.T)
#print('num_1: {}'.format(A_processed.dot(toeplitz(R_clean))))
#print('num: ', numerator)
denominator = A_clean.dot(toeplitz(R_clean)).dot(A_clean.T)
#print('den: ', denominator)
#log_ = np.log(max(numerator / denominator, 10e-20))
#print('R_clean: ', R_clean)
#print('num: ', numerator)
#print('den: ', denominator)
#raise NotImplementedError
log_ = np.log(numerator / denominator)
#print('np.log({}/{}) = {}'.format(numerator, denominator, log_))
distortion.append(np.squeeze(log_))
start += int(skiprate)
return np.array(distortion)
#@nb.jit('UniTuple(float32[:], 3)(float32[:])')#,nopython=True)
def lpcoeff(speech_frame, model_order):
# (1) Compute Autocor lags
# max?
winlength = speech_frame.shape[0]
R = []
#R = [0] * (model_order + 1)
for k in range(model_order + 1):
first = speech_frame[:(winlength - k)]
second = speech_frame[k:winlength]
#raise NotImplementedError
R.append(np.sum(first * second))
#R[k] = np.sum( first * second)
# (2) Lev-Durbin
a = np.ones((model_order,))
E = np.zeros((model_order + 1,))
rcoeff = np.zeros((model_order,))
E[0] = R[0]
for i in range(model_order):
#print('-' * 40)
#print('i: ', i)
if i == 0:
sum_term = 0
else:
a_past = a[:i]
#print('R[i:0:-1] = ', R[i:0:-1])
#print('a_past = ', a_past)
sum_term = np.sum(a_past * np.array(R[i:0:-1]))
#print('a_past size: ', a_past.shape)
#print('sum_term = {:.6f}'.format(sum_term))
#print('E[i] = {}'.format(E[i]))
#print('R[i+1] = ', R[i+1])
rcoeff[i] = (R[i+1] - sum_term)/E[i]
#print('len(a) = ', len(a))
#print('len(rcoeff) = ', len(rcoeff))
#print('a[{}]={}'.format(i, a[i]))
#print('rcoeff[{}]={}'.format(i, rcoeff[i]))
a[i] = rcoeff[i]
if i > 0:
#print('a: ', a)
#print('a_past: ', a_past)
#print('a_past[:i] ', a_past[:i])
#print('a_past[::-1] ', a_past[::-1])
a[:i] = a_past[:i] - rcoeff[i] * a_past[::-1]
E[i+1] = (1-rcoeff[i]*rcoeff[i])*E[i]
#print('E[i+1]= ', E[i+1])
acorr = np.array(R, dtype=np.float32)
refcoeff = np.array(rcoeff, dtype=np.float32)
a = a * -1
lpparams = np.array([1] + list(a), dtype=np.float32)
acorr =np.array(acorr, dtype=np.float32)
refcoeff = np.array(refcoeff, dtype=np.float32)
lpparams = np.array(lpparams, dtype=np.float32)
#print('acorr shape: ', acorr.shape)
#print('refcoeff shape: ', refcoeff.shape)
#print('lpparams shape: ', lpparams.shape)
return acorr, refcoeff, lpparams
class STOI(nn.Module):
def __init__(self):
super(STOI, self).__init__()
self.fs = 16000
self.num_bands = 15
self.center_freq = 150
self.min_energy = 40
self.fft_size = 512
self.fft_in_frame_size = 256
self.hop = 128
self.num_frames = 30
self.beta = 1 + 10**(15 / 20)
self.fft_pad = (self.fft_size - self.fft_in_frame_size) // 2
scale = self.fft_size / self.hop
window = np.hanning(self.fft_in_frame_size)
zero_pad = np.zeros(self.fft_pad)
window = np.concatenate([zero_pad, window, zero_pad])
fft = np.fft.fft(np.eye(self.fft_size))
self.rows = self.fft_size // 2 + 1
fft = np.vstack((np.real(fft[:self.rows,:]), np.imag(fft[:self.rows,:])))
fft = window * fft
self.fftmat = nn.Parameter(torch.FloatTensor(fft).unsqueeze(1), requires_grad=False)
self.octmat, _ = self._get_octave_mat(self.fs, self.fft_size,
self.num_bands, self.center_freq)
self.octmat = nn.Parameter(torch.FloatTensor(self.octmat), requires_grad=False)
def forward(self, prediction, target, inteference):
# pred, targ = self._remove_silent_frames(prediction, target)
# (batch, 1, time) to (batch, fft_size, frames)
pred_mag, pred_phase = self._stft(prediction)
targ_mag, targ_phase = self._stft(target)
# (batch, fft_size, frames) to (batch, frames, fft_size)
pred_mag = pred_mag.permute(0, 2, 1).contiguous()
targ_mag = targ_mag.permute(0, 2, 1).contiguous()
# (batch, frames, fft_size) to (batch, frames, num_bands)
x = torch.sqrt(F.linear(targ_mag**2, self.octmat))
y = torch.sqrt(F.linear(pred_mag**2, self.octmat))
# (batch, frames, num_bands) to (batch, num_bands, frames)
x = x.permute(0, 2, 1).contiguous()
y = y.permute(0, 2, 1).contiguous()
corr = 0
for i, m in enumerate(range(self.num_frames, x.size()[2])):
# segment (batch, num_bands, frames) to (batch, num_bands, new_frames)
x_seg = x[:, :, m - self.num_frames : m]
y_seg = y[:, :, m - self.num_frames : m]
alpha = torch.sqrt(torch.sum(x_seg**2, dim=2, keepdim=True) / (torch.sum(y_seg**2, dim=2, keepdim=True) + 1e-7))
y_prime = torch.min(alpha * y_seg, self.beta * x_seg)
corr += self._correlation(x_seg, y_prime)
return -corr / (i + 1)
def _stft(self, seq):
seq = seq.unsqueeze(1)
stft = F.conv1d(seq, self.fftmat, stride=self.hop, padding=self.fft_pad)
real = stft[:, :self.rows, :]
imag = stft[:, self.rows:, :]
mag = torch.sqrt(real**2 + imag**2)
phase = torch.atan2(imag, real)
return mag, phase
def _get_octave_mat(self, fs, nfft, numBands, mn):
f = np.linspace(0, fs, nfft+1)
f = f[:int(nfft/2)+1]
k = np.arange(float(numBands))
cf = 2**(k/3)*mn;
fl = np.sqrt((2.**(k/3)*mn) * 2**((k-1.)/3)*mn)
fr = np.sqrt((2.**(k/3)*mn) * 2**((k+1.)/3)*mn)
A = np.zeros((numBands, len(f)) )
for i in range(len(cf)) :
b = np.argmin((f-fl[i])**2)
fl[i] = f[b]
fl_ii = b
b = np.argmin((f-fr[i])**2)
fr[i] = f[b]
fr_ii = b
A[i, np.arange(fl_ii,fr_ii)] = 1
rnk = np.sum(A, axis=1)
numBands = np.where((rnk[1:] >= rnk[:-1]) & (rnk[1:] != 0))[-1][-1]+1
A = A[:numBands+1,:];
cf = cf[:numBands+1];
return A, cf
def _remove_silent_frames(self, x, y):
pass
def _correlation(self, x, y):
'''
Input shape is (batch_size, bands, time dimension)
'''
xn = x - torch.mean(x, dim=2, keepdim=True)
xn /= torch.sqrt(torch.sum(xn**2, dim=2, keepdim=True))
yn = y - torch.mean(y, dim=2, keepdim=True)
yn /= torch.sqrt(torch.sum(yn**2, dim=2, keepdim=True))
r = torch.mean(torch.sum(xn * yn, dim=2))
return r